1. Field of the Invention
This invention is related to the scalable encoding of an audio signal and more specifically to methods for performing this data rate scaling in an efficient matter for multichannel audio signals including hierarchical filtering, joint coding of tonal components and joint channel coding of time-domain components in the residual signal.
2. Description of the Related Art
The main objective of an audio compression algorithm is to create a sonically acceptable representation of an input audio signal using as few digital bits as possible. This permits a low data rate version of the input audio signal to be delivered over limited bandwidth transmission channels, such as the Internet, and reduces the amount of storage necessary to store the input audio signal for future playback. For those applications in which the data capacity of the transmission channel is fixed, and non-varying over time, or the amount, in terms of minutes, of audio that needs to be stored is known in advance and does not increase, traditional audio compression methods fix the data rate and thus the level of audio quality at the time of compression encoding. No further reduction in data rate can be effected without either recoding the original signal at a lower data rate or decompressing the compressed audio signal and then recompressing this decompressed signal at a lower data rate. These methods are not “scalable” to address issues of varying channel capacity, storing additional content on a fixed memory, or sourcing bit streams at varying data rates for different applications.
One technique used to create a bit stream with scalable characteristics, and circumvent the limitations previously described, encodes the input audio signal as a high data rate bit stream composed of subsets of low data rate bit streams These encoded low data rate bit streams can be extracted from the coded signal and combined to provide an output bit stream whose data rate is adjustable over a wide range of data rates. One approach to implement this concept is to first encode data at a lowest supported data rate, then encode an error between the original signal and a decoded version of this lowest data rate bit stream. This encoded error is stored and also combined with the lowest supported data rate bit stream to create a second to lowest data rate bit stream. Error between the original signal and a decoded version of this second to lowest data rate signal is encoded, stored and added to the second to lowest data rate bit stream to form a third to lowest data rate bit stream and so on. This process is repeated until the sum of the data rates associated with bit streams of each of the error signals so derived and the data rate of the lowest supported data rate bit stream is equal to the highest data rate bit stream to be supported. The final scalable high data rate bit stream is composed of the lowest data rate bit stream and each of the encoded error bit streams.
A second technique, usually used to support a small number of different data rates between widely spaced lowest and highest data rates, employs the use of more than one compression algorithm to create a “layered” scalable bit stream. The apparatus that performs the scaling operation on a bit stream coded in this manner chooses, depending on output data rate requirements, which one of the multiple bit streams carried in the layered bit stream to use as the coded audio output. To improve coding efficiency and provide for a wider range of scaled data rates, data carried in the lower rate bit streams can be used by higher rate bit streams to form additional higher quality, higher rate bit streams.